webrtc/media/engine/webrtcvoiceengine.cc - Issue 2219653004: Remove old methods in AudioTransport, make it pass a gn build

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Unified Diff: webrtc/media/engine/webrtcvoiceengine.cc

Issue 2219653004: Remove old methods in AudioTransport, make it pass a gn build (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Fix compile. Created 4 years, 4 months ago

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Index: webrtc/media/engine/webrtcvoiceengine.cc

diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc

index be1888eea338392e0f3952061990d8ba24a776b1..afe0975c07755d1882234c7d05e9111bf10f3d37 100644

--- a/webrtc/media/engine/webrtcvoiceengine.cc

+++ b/webrtc/media/engine/webrtcvoiceengine.cc

@@ -1218,12 +1218,9 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream

RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread());

RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread());

RTC_DCHECK(voe_audio_transport_);

- voe_audio_transport_->OnData(config_.voe_channel_id,

- audio_data,

- bits_per_sample,

- sample_rate,

- number_of_channels,

- number_of_frames);

+ voe_audio_transport_->PushCaptureData(config_.voe_channel_id, audio_data,

+ bits_per_sample, sample_rate,

+ number_of_channels, number_of_frames);

}

// Callback from the |source_| when it is going away. In case Start() has

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